This is fairly specific, but I didn’t find any posts out there to help me.
If you are using an older Freepbx with Asterisk 1.4 or 1.6 and you want to include a ZAP or DAHDI channel in a paging group, but it never rings the FXS channel, then you need to examine the dialplan.

I noticed that when it reached this point (- Executing [PAGE203@ext-paging:2] ChanIsAvail), it would hangup.
After tinkering I realized that it was checking on ZAP/4 (my channel in this case).

To fix:
-Go to the extension in Freepbx, and under the “Dial: ” options for that extension, change it from ZAP/4 to DAHDI/4.

Apply and now when you see it doing the Chanisavailable check it will pass and should call your FXS line.

Safe1405 is a bash script that will backup the popular Asterisk directories and place them in a tar.gz file. The script can be used in 1 of 2 modes.
The first mode is interactive, with a menu to choose:

Backup Everything (/etc/asterisk, Voicemail, Recordings, Sounds)

Just VM

Just Recordings

Just Sounds

After the file is created, it will prompt to upload to a remote FTP server.

The second mode is silent, and is best suited for automated execution via cron or at.
Just enable the UNATTENDED and PUT_FTP variables by setting them to “1″. You can directly edit the paths to each directory if yours are non-standard.

I’ve seen several posts around the net from users that want to distribute the calls across multiple trunks with FreePBX. The main reason being that they have a deal with a provider, and each line has 4,000 minutes for free, etc. After looking around at the different methods to accomplish this, it become very clear what must happen.

I’ve had the task of installing GNUDialer for testing and call-centers tend want dialers all the time. Many dialer platforms are extremely bloated and their support fees are enormous! I like GNUDialer because they decided to make an open-source dialer platform that is minimal and functional.

Disclaimer: I am not responsible for any loss of data, server explosions, recessions, depressions, or anything negative that may arise from you running the dialer kickstart script. It was meant with good intentions, but I am not going to be liable for any of your actions. Glad that is out of the way.

You can read through each script that runs and know exactly how the install was done. Make sure you watch the console throughout the install to catch any possible errors that may be thrown out (2)stderr.

2. Start the server with the CD and set the BIOS to boot from CD first

3. You will see this screen if it booted from the CD properly. At the prompt type ‘linux ks=http://www.thiscoolsite.com/scripts/ks_dialer/ks.cfg’ just like it did in the screenshot. Then hit enter.
4. If you typed the command incorrectly you will see this screen asking you to ‘test your media’. If you see this screen reboot your server and try typing the command again, but correctly.

5. After a minute you should see a screen that looks like this. That means you typed the ‘linux ks’ command correctly and it is installing in txt mode. It will take at least 10 minutes, then it will look like it is paused at the end of the install. Don’t worry, it is using yum to install all the latest packages you need to run Asterisk and Gnudialer. The install will reboot itselft, so don’t mess with it.

Upon first boot, the main script runs that will be doing the MySQL databse config, Asterisk install, astcrm, and GNUDialer installs. The script will exit and you will be prompted with a login screen. The default user is ‘root’ and password is ‘star1405′.

Change your root password immediately with the ‘passwd’ command.

Once you are logged in you can check to see that Asterisk is running by typing Asterisk -r and getting in to the console. You can then start the gnudialer daemon for the first time by typing ‘gnudialer –safe’. You should be up and running now! Good Job.

Navigate to http://InternalIPAddress/gnudialer to login to the management interface. Password is ‘gnudialer’.

Make sure you read all of the readme and install files in /usr/src/gnudialer so you can actually setup working campaigns and use the settings properly. Also, a few people are always available for live help in IRC.freenode.net’s #gnudialer channel. Be aware, ignorance is not allowed. If you didn’t attempt to troubleshoot the issue yourself or read everything available to you on the internet, you will be chastised and humiliated in front of your peers. <G>

As always, leave a comment with your email and I will try to help in any way that I can.
Happy automated-dialing trails…to you!
-Gregg

Update 20100222:
From a helpful contributor. Since the encoding has changed and I haven’t had to do file conversion in years and years.
Check out Ernesto’s script here.

I have had many jobs (consulting and Salary) that have involved using Asterisk in a “Call-Center” environment. Whenever an agent would make a sale, the call would need to be recorded for verification purposes. Now I could talk about Asterisk and call-recording for pages upon pages, but I will be focusing on large .wav to .mp3 conversion jobs on the Linux/Asterisk server itself. First off, Asterisk can record calls using several different methods. The first method being that you setup a feature called Automon in /etc/asterisk/features.conf and use *1 (or whatever you specify) to record a call in progress. The second method is recording every call that comes in through a specific DID or enters a specific queue. Another method is to setup a ‘call genie’ that you conference your phone to and it records the entire call-bridge. There are pros and cons to every type of recording method and format. You can use GSM, wav, g729, etc. to record calls. Not matter what method you use, the simple script I wrote will convert your .wav files to .mp3 and also retain the timestamp of when the file was initially created (recorded). The options I use for mlame are what makes this script retain its awesome-ness over time. The options allow the final-result mp3 to be compressed and compatable with Asterisk. This means that you can playback your .mp3 files that you converted over Asterisk if you are using a ‘call-genie’ setup. I have also written scripts to convert your .g729/.gsm/etc => .mp3 so if you need one of those scripts leave a comment and I will email it to you (20090908 UPDATE: Check end of post for .g729 to MP3 file conversion script)

13. Run the script by navigating to the directory and ./recordings_convert.sh

You should see it take a few seconds for each file and go on. I usually test it out by letting it convert a few files then Cntrl + C and ls -l the directory to check the filesizes and timestamps. There is a commented out section at the bottom of recordings_convert.sh that moves the .wav files to a separate directory (easier to check if it worked) and in my case, I keep the old files for a month or so for job security reasons. If you lose a bunch of verification recordings, you are going to have major problems.

Cron it, at it, and happy converting.

__________________________________

Code for .g729 to MP3 script. Very dirty, but works, please test on a small group of test files first.
Leave a comment if you need further assistance.
#!/bin/bash
#Author: Gregg Hansen 20080414
#Used to convert from .g729 -in and -out files to .mp3
#Run on Recordings server

Flip1405 is an Asterisk Failover solution that was created in early 2008. I found the need to create an easy to install failover solution for Asterisk servers. After a couple hours on a Sunday night, I had the first version completed. In 2009 someone who came across this script reworked the code to add new features, and I updated the site. Now, in 2011 I am glad to announce that there is a new version that has many new features and improvements over my existing code. Many thanks to Jonathan Bayer for updating the script again. His site is http://linuxnotes.us

Excerpt taken from his site:
Congratulations! You have now set up your first Asterisk-based VOIP server.

Next question (probably the first your boss will ask you): What happens when the server crashes, or loses power, or needs to be upgraded, or…….; well, you get the idea. If you only have a single server, you lose your phones until it is rebooted. Given that you probably installed this on a cheap, inexpensive piece of hardware, these questions are not out of the question, especially for a business.

We had the same questions, and have implemented a failover solution. I started with a script I got from this web page, but was dissatisfied with it. It seemed to be a bit of a kludge, in that it wasn’t a true HA solution, but was only a backup solution.

I did a massive reworking of the script, and the result downloadable from here:

To use, first get your primary Asterisk server running smoothly. When ready, you can either clone it using Clonezilla (or some other utility) or just install Asterisk on the second box. Then follow these steps:

Setup shared-keys so the servers can copy between each other without user intervention. Setup Login without password
Set up ftp (I use vsftp), if needed. We use it to download the configurations for the Aastra phones. Usually we would use tftp, but since we have some phones outside our office we simply use ftp for everything.
Install this script into /usr/local/sbin on the primary/active system.
Run the script with the -s option:
/usr/local/sbin/flip1405.sh -s
The second script: syncftp.sh, is used to synchronize the /tftpboot directory with the /var/ftp directory. If you need it, install the syncftp.sh script into /usr/local/sbin
Open up the flip1405.sh in an editor, and check the various settings. I would recommend that if you need to change them, to use a config file (/etc/sysconfig/flip1405) rather than editing each file. The config file is NOT sync’d between the systems, so you can have system specific options in the file.
There is no need to specify the device, or whether the system is a master or slave. The script determines this at run time. The most important lines to set are the first three variables: MASTERIP, SLAVEIP, and FLOAT. These are:
MASTERIP IP address of the master system (arbitrary name)
SLAVEIP IP address of the slave system
FLOAT IP address that will float between the Master and Slave, and which your phones will access
All other variables are set for an Elastix system. Elastix is based on the following:
Asterisk 1.6
Centos 5.5
FreePBX 2.6
Make sure to set the bindaddr to the virtual IP address in /etc/asterisk/sip.conf or
/etc/asterisk/sip_general_custom.conf
bindaddr=192.168.1.12 ;IP of Virtual interface
Add the following line(s) to the root crontab (only install the syncftp.sh line if you are using it):
* * * * * /usr/local/sbin/flip1405.sh
*/5 * * * * /usr/local/sbin/syncftp.sh
If you like, you can add the following lines to /etc/crontab instead: